The present invention relates generally to data communication systems. More particularly, the present invention relates to a technique for the flexible implementation of speech coding in a data communication network in a manner that facilitates voice transmissions, data modem transmissions, and facsimile modem transmissions.
The public switched telephone network (PSTN) is primarily used to transmit bi-directional voice calls between two end users. In recent years, particularly with the increasing popularity of the Internet, the PSTN has become a universal system that handles voice calls, data modem calls, facsimile modem calls, and other calls. The data transmission scheme and certain operational parameters may vary from connection to connection, depending upon the specific type of information being transmitted during the current communication session. For example, a relatively low data rate may suffice in the context of conventional voice calls, while a relatively high data rate may be desired for certain data modem applications.
The conventional PSTN is a circuit switched network that transmits data at a rate of 64,000 bits per second (64 kbps). In accordance with well known techniques, the PSTN utilizes pulse code modulation (PCM) techniques to transmit voice calls; ITU-T Recommendation G.711 (November 1988) sets forth the functional requirements for such 64 kbps PCM transmissions. The entire content of Recommendation G.711 is incorporated by reference herein. Although voice calls can be adequately handled via the G.711 scheme, a number of speech coding techniques may be employed to conserve bandwidth resources associated with the transmission of voice calls without sacrificing the quality of the voice transmission. For example, ITU-T Recommendation G.726 (December 1990) sets forth an adaptive differential pulse code modulation (ADPCM) scheme that may be used in the context of voice calls. The G.726 ADPCM technique effectively reduces the normal PSTN data rate from 64 kbps to 40 kbps, 32 kbps, 24 kbps, or 16 kbps. As another practical example, ITU-T Recommendation G.728 (September 1992) sets forth a speech coding methodology that effectively reduces the normal PSTN data rate to 16 kbps using linear prediction techniques. Other speech coders that may be utilized in this context (and in the context of the present invention) are set forth in ITU-T Recommendation G.723.1 and ITU-T Recommendation G.729a. The G.723.1 Recommendation contemplates a 5.3 kbps data rate and a 6.3 kbps data rate, while the G.729a Recommendation contemplates an 8 kbps data rate. The entire contents of Recommendations G.726, G.728, G.723.1, and G.729a, along with any related appendices, attachments, and summaries, are incorporated by reference herein.
The PSTN rarely uses ADPCM or other speech coding techniques because such techniques can be detrimental to data and facsimile modem transmissions. Although ADPCM-encoded speech sounds essentially identical to PCM-encoded speech, modem signals, both fax and data signals, are sensitive to the coding effects of ADPCM and suffer performance degradation compared to operation over G.711. Consequently, the performance of a modem system suffers in the context of an ADPCM channel; the data rate may be reduced by up to fifty percent. Thus, in response to the universal application of the PSTN as a conduit for voice, facsimile, and data calls, and in an effort to satisfy modem users seeking high data rates, the traditional 64 kbps PCM scheme is often employed as the primary transmission protocol within the PSTN.
In view of the circuit switched nature of PSTN calls, conventional data communication systems are not able to determine, a priori, whether a new call conveys voice information, data modem information, or facsimile modem information. Accordingly, G.711 is conventionally used when a call is established in the PSTN regardless of whether the call is a voice call or a modem call. Furthermore, the PSTN architecture is configured such that the circuit is fixed for the duration of the call. Consequently, if the call is a modem call and ADPCM encoding is present in the communication channel, then the modem transmission speed will inevitably suffer. Conversely, if the call is a voice call and speech coding is not utilized in the channel, then network resources, e.g., bandwidth, may be wasted.
Prior art data communication systems may handle some intercontinental facsimile calls in a manner that addresses the effects of speech coding. This technique is commonly referred to as demod/remod. The demod/remod technique is typically utilized for facsimile transmissions across communication channels, e.g., submarine cables, that employ ADPCM encoding. As discussed above, facsimile transmissions may be adversely affected by an ADPCM channel. In accordance with the demod/remod scheme, the call is initially established such that the end devices can transmit their respective call and answer signals. If a facsimile calling tone is detected, then intermediate facsimile modems are xe2x80x9cinsertedxe2x80x9d into the connection at the ends of the ADPCM channel. Notably, the activation of the intermediate facsimile modems occurs before the end devices begin any training procedures. The intermediate facsimile modems are configured to communicate with each other over the ADPCM channel and with the respective end devices. Accordingly, the calling side intermediate modem informs the answer side intermediate modem that a facsimile call is being placed to a certain telephone number.
Once the facsimile modem connection is initialized between the answer side intermediate facsimile modem and the destination modem device, the calling side intermediate facsimile modem establishes a compatible connection with the originating modem device. Thus, the facsimile data is transmitted between the originating modem device and the calling side intermediate facsimile modem (and between the answer side intermediate facsimile modem and the destination modem device) at a relatively high data rate in accordance with the G.711 protocol. In contrast, the facsimile data can be transmitted over the ADPCM channel at a relatively low data rate, e.g., 9.6 kbps, in accordance with conventional facsimile modem methodologies. In this manner, the ADPCM channel resources may be allocated in a more efficient manner.
Although the demod/remod technique can be employed to reduce the amount of traffic across an ADPCM link, the technique only works in connection with facsimile modem transmissions because facsimile calls can be identified by a distinct calling tone. In addition, the demod/remod technique can add a significant amount of delay to the connection time and to the overall facsimile transmission time. Notably, the demod/remod procedure is not applicable to data modem calls, where a high data rate (e.g., a rate higher than the maximum rate attainable for an ADPCM channel) is often desired.
A data communication system according to the present invention is capable of flexibly configuring itself to handle voice calls, data modem calls, and facsimile modem calls in a manner that conserves network bandwidth resources. The data communication system detects the type of information to be conveyed by the current call such that an appropriate data transmission scheme can be utilized for the current call. In a practical embodiment, a 64 kbps transmission protocol, such as the G.711 PCM protocol, may be used for data modem calls to facilitate a relatively high data rate. In contrast, speech coding techniques may be utilized for voice calls to conserve bandwidth resources associated with the communication network. Unlike some prior art methodologies that are limited to facsimile applications, the techniques of the present invention can be employed to dynamically alter the data transmission scheme for voice calls and modem calls.
The above and other aspects of the present invention may be carried out in one form by a data communication system having a calling device, an answer device, and a transmission network capable of dynamically allocating bandwidth resources for purposes of transmitting calls. The transmission network includes an interface component configured to convert from a first data coding technique associated with the calling device to a second data coding technique utilized by the transmission network. The data coding conversion is performed during the initialization of a call and is performed in response to the type of call, e.g., a voice call, a data modem call, or a facsimile modem call.